Friday, April 28, 2017

I'm
planning to write a piece of music for a sound installation using Logic
Pro. There would be 10 tracks, which I'm hoping to route from an audio
interface such as a MOTU 828 to 10 different active speakers, so that
there would be one speaker for each track of the piece. I'm quite new to
music technology, so I'd appreciate it if you could explain some things
to me. Firstly, would this setup actually work? Secondly, is a MOTU 828
a good device to use or would something else be better? Also, what
leads do I need, what active speakers would you recommend for use
outdoors, and how should I protect the speakers and other equipment from
getting wet, without compromising the sound?

I have around £1500 to spend on speakers, audio interface and leads, and I would really appreciate your help.

Anya Ustaszewski

This
system diagram shows the plan of an eight-channel sound installation in
a fairly large church. Each of the four CD players was sending a
split-mono signal to dedicated amplifiers, which were connected via
two-core mains cable to speakers around the building. The result was,
well, spacious!

News Editor Chris Mayes-Wright replies:
In response to your first question, the answer is yes, it will work; in
fact, I've done a number of similar projects myself. The MOTU 828 that
you mention would indeed be a suitable device, but it only has eight
analogue outputs. Analogue outputs are the kind that you can connect
directly to the line inputs on the back of an active speaker. So,
basically, if you used the MOTU 828 on its own (without additional
equipment), you would only be able to feed eight different signals to
eight different speakers. If doing this, you should use quarter-inch
balanced jacks to connect between the back of the MOTU 828 and the
inputs to your active speakers.

If you really need 10 channels, you can use the MOTU 828's stereo
S/PDIF output: this is a digital audio connection that needs converting
to analogue before it can be plugged into the back of a speaker. You can
buy digital-to-analogue (D-A) converters from most music technology
equipment retailers, such as the ones that advertise in these pages, but
they can be quite expensive. With your budget, I'd suggest aiming at
the lower end of the market. Behringer manufacture a product called the
Ultramatch Pro SRC2496, which has D-A functionality, as well as some
other features. It costs £109 including VAT, so it shouldn't take up too
much of your budget, but if you can make do with just eight speakers, I
would suggest using the MOTU interface on its own.

You mentioned that you'd like to use active speakers. When I've done
installations in the past, I've found that using passive speakers with
separate power amplifiers was far easier (and safer). I'm a big fan of
the JBL Control 1 speakers (they cost under £100 for a pair and are
really rugged), which seem to work well with Samson Servo 170
amplifiers, although these seem to have been superseded by the Servo 200
model, which costs around £100.

The reason I suggest passive speakers coupled with amps is that if
you are using active speakers and they're spread out over a wide space,
you will need to distribute power to each of them, which is not a good
idea over distance, especially outdoors. I would only suggest active
speakers if you can guarantee that there will be a power socket next to
each one.

If you use passive speakers powered with amplifiers, you can keep all
the amplifiers (and therefore power) in one place, then run long
lengths of speaker cable to each one. Mains electricity cable works well
to carry the signal between the amplifier and speaker, as its per-metre
cost is cheaper than 'proper' speaker cable, and it's tough enough to
withstand a bit of trampling in grass or on concrete.
Remember that you only need one stereo amplifier for every two
speakers, because a stereo amplifier has two separate inputs and
outputs. As long as you feed a dual mono signal into the amp, you will
get the necessary separation at the speakers.

You also mention that you are planning on running all of this from a
laptop. If so, bear in mind the security aspect; it's probably not wise
to leave a laptop unattended at a public event. Also, the fact that the
audio would be running from a hard drive (which is not the most stable
of media) may be a concern.
If these things worry you, and you're not too bothered about the sync
of the material being ultra-tight, I would suggest using a bank of CD
players; these can be located near the amplifiers to help keep the power
supply in one place. I have used CD players for exactly this purpose in
the past, and it is by far the easiest and most reliable way to play
out sound for a long period of time.

Again, as with the amplifiers, you only need one stereo CD player for
each two channels of your sound installation. But you'll need to bounce
five stereo mixes (or four, if you're only using eight speakers) from
Logic, then burn them onto CD. Previously, I've used the Tascam CD160,
but any CD player with line outputs (usually red and white phono
sockets) would be fine. The cabling for connecting the CD players to the
amplifiers can be done using phono-to-jack connectors.

I found the Tascam player to be particularly good at this job: it can
be rackmounted, it's pretty robust, and if you have a stack of them the
infra-red remote control can be used to start them off simultaneously,
to achieve a reasonably good sync between them! Of course, you could get
a few of your friends to press the play buttons at the same time, but
this approach is a little rudimentary.

As for your concerns about the weather, the most important thing is
to run everything through a safety breaker circuit, so if you get a
problem the power instantly cuts and everything is made safe.
To prevent the rain getting to your speakers, you can cover them with
plastic refuse sacks. Also, make sure all electrical contacts are
insulated with electrical tape, and if you are using a stack of power
amps, ensure that these are well covered, while keeping vents, which
require a constant airflow around them, clear. Make sure you lift the
equipment off the ground, too (perhaps on a table or wooden pallet), and
erect some kind of cover, such as a garden gazebo or large patio
umbrella, over it all. Good luck!

Tuesday, April 25, 2017

Referring to your article in Q&A from SOS
January 2007, regarding a basic home recording setup, can you connect
the Yamaha MG102C, which you suggest, to a PC directly? If so, how?

The
output section of a mixer may have a number of different connectors
(this Yamaha MG102 only uses jack sockets), so you'll need to have the
correct cable to interface with the PC input. Via email

News Editor Chris Mayes–Wright replies:
To get the audio from a mixer (or any source) directly into a
computer, you'll need some kind of audio interface — also known as a
soundcard. It's likely that your computer has an 'on–board' soundcard,
and yours may have line and/or microphone inputs on mini-jack
connections. If this is the case, you can simply plug the outputs of the
mixer (in this case either the Stereo Out, Monitor, Rec Out or Aux
Send) into the input of your on–board soundcard. You'll obviously need a
cable with the relevant terminations. If connecting from the
quarter–inch jacks of the MG102C to a mini–jack line input of your PC
soundcard, you'll need two mono quarter–inch jacks on one end of the
cable and a stereo mini–jack on the other. From here, almost any audio
recording package should be able to 'see' the incoming audio.

An alternative would be to purchase an external audio interface, and
you can get these from around £50. Not knowing your specific
requirements, I can't really recommend one that will best suit your
needs, so I suggest visiting our on–line Forum, www.soundonsound.com/forum, where you can post questions.

Saturday, April 22, 2017

I don't really want to get into a discussion about
whether or not you should mix on headphones, but I'm wondering whether
I'd be better off with a pair of Sennheiser HD600 (or similar)
headphones instead of some low–end monitors? At the moment my 'studio'
is just the corner of my bedroom and there's a huge peak around 150Hz,
so it's not ideal for mixing. My current monitors are Blueroom Minipods,
which are not the best thing for monitoring. So most of the time I mix
on Sennheiser HD497s and check later on the monitors.

Sennheiser
HD600s are well respected, and are often used for critical listening.
They'll yield better results on a mix than a pair of poorly placed or
indifferent–sounding speakers, particularly if your listening
environment is compromised. There are certain considerations to bear in
mind when mixing on headphones, however.

The problem is
that my mixes are not really as good as I would like them to be, so I'm
just wondering what the next step should be. (I don't think the missus
wants bass traps on top of her bed, by the way...) So, basically, the
question is: what monitors should I be looking for in order to have
something better than mixing on, say, a pair of HD600s and checking on
the Minipods?

SOS Forum Post

SOS contributor Mike Senior replies: I'd rather mix
on a good pair of headphones than a similar–priced pair of (especially
ported) speakers any day of the week, especially in a dodgy–sounding
room. The Sennheiser HD600s are a very well–respected choice in this
regard, and should seriously outclass your HD497s, in my opinion.
However, the way that headphones place phantom central images 'inside'
your head, and the lack of crosstalk between left and right channels (as
in loudspeaker listening), can make it a bit tricky to judge panning
and balance between the instruments. Gauging the low–end response will
also be tough, as headphones simply can't generate the same physical
sensation as the low frequencies emerging from loudspeakers.

A little temporary acoustic treatment should allow you to use your
existing monitors to judge panning and stereo imaging, so working around
that aspect of headphone mixing shouldn't be too much of a concern.
However, in your position I'd probably invest in an additional mono
'grotbox' (such as the Avantone Mixcube or Pyramid Triple P, perhaps),
which should give a more reliable impression of the balance of the
instruments than either the headphones or Minipods are likely to provide
— it won't sound nice, but that's not the point of it — and will also
confirm mono compatibility.A
single 'grotbox', such as this Avantone Mixcube, will give a better
idea of musical balance than headphones or unsuitable monitors.
'Grotboxes' are useful in all studios, large and small.

The
bass problems are more difficult to work around, however. The HD600s
are very good in this respect (especially if you make sure to reference
your mix against commercial tracks), and could be supplemented with a
software spectrum analyser (such as Roger Nichols' Inspector), but I'd
still advise checking the low–end response somewhere else if possible.
Even spending lots of money on new monitors won't help you here, if your
room sounds rubbish at the low end. If you can't check mixes elsewhere,
sources with mainly high–frequency and mid-range content, such as
guitars, vocals and drum overheads, can be high–pass filtered. Try,
also, to keep a tight grip on the dynamics of remaining low frequencies.
For more headphone mixing tips, see the 'Mixing On Headphones' feature
in SOS January 2007, which includes links to plug–ins that simulate
loudspeaker listening on headphones.

Thursday, April 20, 2017

Some
music applications will completely fail to take advantage of the
multiple cores of a modern CPU - but which ones, and why? We find out,
and advise on how you can make best use of however many cores your PC
has.

Over
the last couple of years, the PC musician has been offered first
dual-core processors, then quad-core models, and octo-core machines
(currently featuring two quad-core processors) are now available for
those with deep enough pockets. Competitive pricing has already ensured a
healthy take-up of DAWs based around a quad-core CPU, yet many users
haven't cottoned onto the fact that not all software benefits from all
these cores. Some existing software may only be able to use two of them,
reducing potential performance by a huge 50 percent, while older
software may only be able to utilise a single core, reducing potential
performance to just 25 percent of the total available. This month PC
Musician investigates which audio software works with dual-core,
quad-core PCs and beyond, what benefits you're likely to get in practice
over a single-core machine, and which software may for ever languish in
the doldrums.

In the days when most musicians ran Windows 95, 98 or ME, the
question of running multiple processors didn't arise, because none of
these operating systems supported more than a single CPU. It was Windows
NT and then Windows 2000 that introduced us to the benefits of being
able to share the processing load between multiple CPUs: Windows 2000
Professional supported one or two processor chips, while the more
expensive Server version supported up to four, and the Advanced Server
up to eight. However, at this early stage each processor was a
physically separate device, so to be able to (for instance) use twin
processors, you needed a specially designed motherboard with two CPU
sockets. Many audio developers and interface manufacturers didn't
actively support Windows 2000, so most musicians stuck with Windows 98.
In 2001, Microsoft released Windows XP in Home and Professional
versions, and once again most consumers who opted for the Home version
were limited to a single physical processor, although the Professional
version supported two. By this stage many musicians were straining at
the leash, wanting to run more and more plug-ins and software
instruments, and this Professional version let them do exactly that,
using dual-processor motherboards and twin Xeon or Pentium 4 processors.When
you're running stereo audio editors (such as Wavelab 6, shown here) and
stand-alone soft synths or samplers, and even in most multitrack
sequencers when you're only running a single track, only one core of a
multi-core CPU will be heavily used, although any others available may
help with disk access, the user interface and other applications that
are running simultaneously.Multi-processing options really
opened up the following year, when Intel introduced first Xeon and then
Pentium 4C processor ranges with Hyperthreading technology, which let
these CPUs appear to both Windows XP Home and Professional (or Linux
2.4x) as two 'virtual' processors instead of one physical one. They each
shared the various internal 'sub-units', including the all-important
FPU (Floating Point Unit), but could run two separate processing
'threads' simultaneously.

Intel claimed up to a 30 percent improvement with specially written
applications over a standard processor, but as many musicians soon
found, having a Hyperthreaded processor didn't necessarily benefit them
at all unless they were running several applications simultaneously,
since applications like MIDI + Audio sequencers had to be rewritten to
take advantage of Hyperthreading. Steinberg's Nuendo 2 was one of the
few music apps to support it, but although various others followed, a
few (such as Tascam's Gigastudio) needed a major rewrite before they
would even run with HT enabled. Nevertheless, my own tests (published in
PC Notes June 2004) showed that with optimised audio applications such
as Cubase SX2 you could expect a significant drop in CPU overheads where
it really mattered, at low latencies of 3ms or under.

The biggest change came in late 2004, when both AMD and Intel seemed
to agree that processor clock speeds had reached a ceiling. Intel
abandoned plans to release a 4GHz model in their Prescott CPU range, and
in 2005 both companies largely switched to releasing dual-core models.
Unlike the twin virtual processors of Intel's Hyperthreading range,
these featured two separate processing chips mounted inside one physical
package. By placing two processor cores into a single piece of silicon,
manufacturers could provide significantly faster performance than a
single processor, even when under-clocking them and running them at
lower voltages, so that they didn't run hotter than the single-core
variety.

By late 2006 we had been introduced to quad-core processors, which
have now dropped in price and can even be run with Windows XP Home
(which is licensed to run a single physical processor, however many
cores it has inside). However, if running XP Professional (and the x64
64-bit version), Vista Home Premium, Business, Enterprise or Vista
Ultimate you also gain the option of installing two quad-core processors
on a suitable motherboard, to provide a total of eight processing
cores. Unfortunately, as with so many new hardware advancements, much
software has had a long way to catch up before it could take advantage
of so many cores.

Determining how much extra performance you'll get from a particular
software application with four or more cores will require some benchmark
testing, but fortunately it's far easier to determine whether or not a
particular application is utilising all the available cores. Windows
Task Manager (launch it using the Ctrl-Alt-Delete keyboards shortcut,
aka the 'three-fingered salute') has a Performance page that offers a
CPU Usage History, and as long as you select the 'One Graph Per CPU'
option in the View menu you'll get as many individual graphic windows
displaying CPU activity as you have cores.

This
is what you're hoping to see in Task Manager's CPU usage history when
running a multitrack sequencer: the four cores in this PC are all being
equally stressed to their maximum capability.

When you're using a PC with multiple processors of whatever type, to
gain any significant performance benefit the software you run has to be
specially written or adapted with multiple processors in mind. The way
multi-processing works is that applications are divided into 'threads'
(semi-independent processes that can be run in parallel). Even with a
single processor there are huge advantages in this programming approach.
Many applications use multiple threads to enable multi-tasking, so that
one task can carry on while another is started; and when multiple
processors are available, different threads can be allocated to each
CPU.Here's
some classic multi-processing confusion, illustrated by Cubase 4
running 28 voices of a heavy-duty physically modelled soft synth that
together consume almost 100 percent of a single core of this dual-core
PC. The high Cubase 'VST Performance' meter reading (bottom left) simply
indicates that one or more cores is approaching its limit. However,
since Cubase 4 is optimised for multi-processing, if you create another
track and connect its output to another instance of the same soft synth,
you'll still be able to run a further 28 voices on the other core.

With
some processor-intensive programs, such as 3D graphics and CAD
software, it's comparatively easy to split off different functions to
each processor. However, the situation becomes somewhat more complicated
with an application such as a MIDI + Audio sequencer, since all the
different tracks are generally being streamed in real time and must
remain in sync.

Early schemes used by audio software for sharing tasks between
multiple processors were fairly crude; they tended to devote each CPU to
a specific duty, so that (for instance) audio mixing and effects were
handled in one thread, MIDI processing in another, and user interface
responses in yet another. When a MIDI + Audio sequencer is run with
several identical processors under such a scheme, the entire
audio-processing workload is normally handled by one processor, with any
remaining tasks left to the others. Since audio processing is by far
the most significant overhead for any music application, this approach
resulted in a typical overall performance improvement of just 20 to 30
percent for a dual-core processor over a single-core processor running
at the same clock speed.

To gain further improvement, you need to split the audio processing
in some way between the various CPUs, so that it can be processed in
parallel. This means added code and complexity, and rather explains why
some audio software really benefits from four or more cores, while some
doesn't. Steinberg introduced their 'Advanced Multiple Processing
Support' on Cubase VST version 5, splitting the audio processing between
the processors and giving much larger performance boosts of 50 to 60
percent. Many other audio developers (although not all) followed with
similar improvements, and although there are no guarantees, most
applications optimised in this way should also subsequently benefit from
quad-core and octo-core PCs.

Despite the possibilities, even today many mainstream office
applications and games have not really been optimised for multiple
processors, and some developers have been resistant to rewriting their
applications to support more than two cores, since debugging an
application that can run several threads in parallel is far harder than
one in which everything happens in a single queue of tasks. Of those
applications that have been optimised for multiple processors,
most can still only take advantage of two processors, so you'll only get
the best performance from them on a dual-core or twin single-core
computer. If, for instance, you run a game that can only take advantage
of two cores on a quad-core machine, it will only be able to access up
to 50 percent of the available processing power.

With quad-core processors and beyond, applications that may benefit
include 3D graphics modelling, ray-tracing, and rendering, plus
video-encoding tasks, image processing and some scientific tasks. You're
always likely to achieve good performance when running several
different applications simultaneously, since each will get a good share
of the pie, but with MIDI + Audio applications you want a single
application to have all its tasks shared out as fairly as possible
between the available cores.

While
it's possible to specifically assign each Windows programming task to a
separate processor, you can also let Windows handle its CPU resources
dynamically across a single processor by giving each task a specific
priority. The lowest priority is nearly always given to the user
interface, which is why screen updates can get sluggish on a single-core
machine when you run lots of real-time software plug-ins.
Conversely,
any PC with multiple cores is always likely to remain more responsive
even when most of the cores are stressed, because the user interface is
still happily ticking away on another one. Even if you're running
elderly applications that are not multi-threaded, you can still benefit
from a dual-, quad- or octo-core machine if you're running several such
applications simultaneously, as Windows will allocate each one to a
different core.

Developers told me that although most instruments and plug-ins run as
several threads, they have no control over how these are distributed
among the available cores. This is totally managed by the host
application, and according to all the tests I carried out while
researching this feature, most audio applications treat each mono/stereo
audio track (or soft-synth/sampler track), plus associated plug-in
effects, as a single task, and allocate it to a single processor core.
You can easily confirm this for your own applications using Task Manager
(see the 'Checking Your Tasks' box) and systematically adding a series
of demanding plug-ins to the same audio track. I suggest a convolution
reverb with the longest Impulse Response you can find.The
older Thonex benchmark masks the true performance of systems with four
cores and beyond, here displaying a very modest performance increase of
no more than 30 percent between identically-clocked dual-core and
quad-core systems.These
DAWbench Blofelds DSP40 results illustrate a much healthier scaling
from two cores to four, and although the number of Magneto plug-ins
isn't the whole story (there are a smaller number of Dynamic and EQ
plug-ins also being run) it nevertheless shows that the small extra cost
of a quad-core CPU over a dual-core model can give you almost double
the performance — as long as the load is shared well between the cores.

If
you're running multiple cores (whether in the same chip or spread
across multiple processors in discrete packages) the above has certain
implications. Let's say you have a physically-modelled synth that
consumes a lot of CPU resources. Since our synth track is a single task,
on a quad-core processor it can only consume a maximum of 25 percent of
the overall processing power available — ie. the maximum available from
a single core. So, even though your sequencer's 'CPU meter' may
indicate 100 percent loading in this situation, and it's possible for
your audio application to glitch and stop playback because one of the
four cores has run out of steam, you still have 75 percent of your CPU
resources available to run other synths and plug-ins, which should
automatically get allocated to the remaining cores. Confusing, isn't it?
So if you find yourself 'maxing out' a single core by, for instance,
running lots of instruments on different tracks, all linked to a single
multitimbral software sampler, launch another instance of it and run
some of your instruments from that one instead.

When measuring multi-core performance of audio applications, it's
therefore important to choose a suitable benchmark test that will allow
the applications the best chance of spreading the processor load as
evenly as possible. I carefully tested single-, dual- and quad-core PCs,
all having identical clock speeds, with Cubase SX running the Thonex
and Blofelds DSP40 tests. As you can see from the graphs, while the
older Thonex test only displays a 20 to 30 percent improvement between
the dual-core and quad-core results, Blofelds showed much better
scaling. A quorum of DAW builders seem to have agreed that Vin
Curiglianio's DAWbench suite is currently the best test available to
measure differences in multi-core system performance, since it starts
with a real-world song and then ignores the application's CPU meter in
favour of adding more and more plug-ins and/or soft synths across 40
tracks until you hear audio glitching, which largely mirrors what many
musicians do in the real world.

The original DAWbench Blofelds DSP40 test is for those Cubase/Nuendo
users who mainly record audio tracks and use lots of plug-ins (there's
now also a new SONARbench DSP test that uses the same techniques), while
the L-Factor II test is for Cubase/Nuendo owners who instead run lots
of software synths. Such 'on the edge' tests are also useful in
comparing audio driver performance, as well as spotting operating system
issues such as jerky graphic scrolling under stress, and the extra
overheads imposed by the Windows XP and Vista Aero graphics over the
Windows 'classic' look.

What tasks you're going to perform with your audio application may
also affect the ideal number of cores, and thus which is the 'best' PC
for the job. For drummers and vocalists monitoring their own live
performances on headphones, the Holy Grail is to run a system that runs
with barely discernible latency. Many would be happy using a buffer size
of 64 samples, which would mean a total real-world latency for audio
monitoring with plug-ins of just under 5ms (at a sample rate of
44.1kHz), or around 3.5ms for playing soft synths. If you still find
this unacceptably high and prefer not to rely on 'zero latency'
monitoring solutions (which bypass any plug-in effects), 32-sample
buffers would offer total audio monitoring latency of around 3.5ms
(around 2.7ms for soft synths), again at a 44.1kHz sample rate.

Blofelds DSP40 tests by a range of DAW builders who have access to
lots of PCs based around different processors have shown that at really
low buffer sizes, such as 32 samples, a single quad-core processor will
always outperform a single dual-core processor or (more interestingly) a
system featuring two dual-core processors, and sometimes even a dual
quad-core system. In some tests at these really low latencies, when
stressed with lots of plug-ins and instruments, the single quad-core
machine was the only one to complete them successfully, making it the
current king for low-latency performance.

If you're happy to run use a higher buffer size, of 128 samples or
above (audio monitoring latency of around 8ms), you'll probably be able
to run significantly more plug-ins and soft synths using two quad-core
processors than one. Those involved in lots of recording work who want
'real time' monitoring may thus prefer a single quad-core, while others
who rely mainly on samples and soft synths may get even more mileage
from a twin quad-core system.

This is the biggie: it's all very well having a hugely powerful
quad-core or octo-core PC, but not a lot of use if your software only
uses two or four cores from those available, or makes a poor job of
sharing resources between them. The secret is for the application to
balance requirements across the available cores, so that you don't get
any audio glitches as a result of one or more cores running out of juice
while there's some still available from the others.

For the reasons mentioned above, stereo audio editors may not take
full advantage of a multi-core PC — something I soon confirmed with
Steinberg's Wavelab 6, which only used one core for DSP processing
during playback or audio rendering. Its author Philippe Goutier says
that a second core will be used for disk access and the user interface,
which does at least mean that the application will always remain
responsive to new commands, but he hopes to improve core-sharing now
that so many musicians have multi-core PCs.
The vast majority of stand-alone soft synths also seem to mostly use a
single core, but as soon as you load the VSTi or DXi version into a
host VSTi or DXi application, this host should distribute the various
plug-ins and soft synths across the available cores to make best use of
resources. Fortunately, most multitrack audio applications can
distribute the combined load from all your tracks between as many cores
as they find, although it's perhaps inevitable that since many of the
latest versions were released long before quad-core and octo-core PCs
were in regular use, some don't manage it quite as efficiently as
others. Even now some developers don't have octo-core test systems.

Before coming to any conclusions about the multi-core performance of
your particular sequencing package, make sure you have any appropriate
parameters set correctly. For instance, in the case of Cubase/Nuendo
you'll need to tick the 'Multi Processing' box in the Advanced Options
area of the Device Setup dialogue, while for Sonar the tick-box labelled
'Use Multiprocessing engine' is the one to check. With these settings
deactivated you'll only be using one of your cores, and performance will
plummet. In Reaper, most multi-core users will need to tick the the 'FX
render-ahead' option in the Audio Buffering dialogue to enable the full
benefits of native plug-in multi-processing. Universal Audio UAD1
owners should leave this option un-ticked, however, because of current
UA driver issues.

Audio
applications (such as Cubase 4 and Reaper, shown here) tend to have
specific tick-boxes to allow you to enable multi-processing support, so
make sure these are activated if you want to achieve the best
performance.

Reaper's Justin Frankel told me that he routinely does a lot of his
development on a dual quad-core Xeon PC, so it's hardly surprising that
the default Reaper settings work well with up to eight-core machines,
typically offering over 95 percent utilisation of all eight cores.
Reaper mostly uses 'Anticipatory FX processing' that runs at irregular
intervals, often out of order, and slightly ahead of time. Apparently,
there are very few times when the cores need to synchronise with each
other, and using this scheme he can let them all crank away using nearly
all of the available CPU power. Exceptions include record input
monitoring, and apparently when running UAD1 DSP cards, which both
prefer a more classic 'Synchronous FX multi-processing' scheme.

Steinberg's Cubase SX, Cubase 4 and Nuendo all work decently on
quad-core systems, scaling up well from single to dual-core and
quad-core PCs. However, Cubase 4 and Nuendo 4 don't currently provide
all the benefits they could at low latency with a dual quad-core system.
Compared with the potential doubling of plug-in numbers from dual to
quad, when you move to 'octo' you may only be able to run about 40
percent more plug-ins down to buffer sizes of 128 samples, while below
this you may even get worse performance than a quad-core system.

Steinberg developers have already acknowledged the problem, which is
apparently due to "a serialisation of the ASIO driver, which eats up to
40 percent of the processing time. Together with the other
synchronisation delays, only 25 to 30 percent of the 1.5-millisecond
time-slice can be used for processing. This is not very efficient."
Steinberg have promised to address the issue in a Nuendo 4 maintenance
update, and have hinted that it may also result in changes to the ASIO
specification.

Cakewalk's Sonar does seem to scale well, sometimes giving a better
percentage improvement when moving from a quad-core to an octo-core PC
than the current version of Nuendo/Cubase 4, but the jury still seems to
be out on whether choosing ASIO or WDM/KS drivers gives better results;
with some systems ASIO is a clear winner, while in others WDM/KS
drivers move significantly ahead.
Digidesign have a reputation for being slow but thorough when testing
out new hardware to add to their 'approved list', and as I write this
in early November 2007 their web site states that Intel Core 2 Quad
processors and Intel Xeon quad-core have not been tested by Digidesign
on Windows for any Pro Tools system.

Nevertheless, Pro Tools HD/TDM users started posting recommendations
for rock-solid systems featuring twin dual-core Opteron processors (four
CPU cores in all) in mid-2006, and there are now loads of Pro Tools LE
users successfully running both quad-core and even a few octo-core PCs
in advance of any official pronouncements (there's lots of specific
recommendations on both quad-core and octo-core PC components in a vast
126-page thread on the Digi User Conference at http://duc.digidesign.com/showflat.php?Cat=&Number=988224).
Despite the lack of official 'qualification', all Pro Tools systems
seem to scale well on quad-cores, happily running all four cores up to
100 percent utilisation, and many users are very pleased with their
quad-core 'native' CPU performance.

Like various other audio applications, even the latest Mac version of
Logic Audio doesn't yet fully benefit from having eight processor cores
at its disposal, but for die-hard PC users of Logic the situation is
rather more serious: Apple discontinued development and support for
those using Logic on the PC back in 2002, so most recent version (5.5.1)
is now some five years old. Although it's a multi-threaded application,
Logic 5.5.1 for Windows is not really optimised for multiple
processors, so only one of the cores is likely to get much of a workout.
However, there's a partial workaround, using the I/O Helper plug-in
available from Logic version 5.2 onwards, which can force any plug-ins
on a track with it inserted to run on a second core, so that you can use
lots more plug-ins/instruments overall (there's a more detailed
description on Universal Audio's web site at www.uaudio.com/webzine/2003/may/index5.html). Logic Audio 5.5.1 also has a problem if more than 1GB of system RAM is installed (see http://community.sonikmatter.com/forums/lofiversion/index.php/t8032.html
for some suggestions on this one), and also has problems running some
VST plug-ins. It's unlikely to benefit from a quad-core processor at
all, and I wouldn't recommend running it on a new quad-core PC, so its
shelf-life is looking increasingly limited.

Overall, getting the best out of a multi-core PC generally means a
little detective work from the user. You need to make sure you have the
most appropriate audio application settings (which might be different if
you run DSP cards), and you also need to be cautious when running
heavy-duty synths or plug-ins that might consume one of your cores in a
single gulp. Keeping an occasional eye on the Windows Task Manager may
also help, since the CPU meters provided by most sequencers are becoming
rather less useful now that they are monitoring so many individual
cores.

Wednesday, April 19, 2017

I recorded a performance of Handel's Messiah
recently, and I was wondering what the common practice is when
compressing the whole mix in classical recordings. During the session, I
put a compressor over the master outputs, just to catch any stray
peaks, and when fiddling around with the settings I found that a low
threshold and a low ratio helped to blend the mix as a whole and
fattened it up. Is this acceptable?

SOS Forum PostRecordings
of classical music typically have an incredibly wide dynamic range. In
this example waveform (above), for instance, the audio level is
generally very low, except for the explosive section about
three-quarters of the way through. Sometimes the best way to reduce this
dynamic range is to manually ride the fader on mixdown, or draw in
volume automation to drop the level of the loudest parts by a few
decibels. Once you've done this, you can make up lost gain by turning
the signal up at the output stage.

SOS contributor Mike Senior replies:
The question is what you're trying to achieve. With a piece like this,
which has a wide dynamic range (between the quieter Recitatives and the
full-scale Hallelujah chorus), I'd certainly recommend reducing the
dynamic range a little to make the CD more suitable for home listening.
The most transparent way of doing this would be to use simple fader
automation, riding up the quieter sections to make them more audible. I
wouldn't go for much more than about a 6dB increase to the quietest
sections if you're unsure how far to go. The advantage with this
approach is that a human engineer can intelligently anticipate changes
in the signal in a way that no compressor can.

Another thing you can also deal with using fader automation (or even
audio editing) is ducking any brief signal peaks which are unduly loud,
which allows you to achieve a louder final CD. Some might suggest
limiting or even soft-clipping to achieve a similar effect, but neither
will sound as transparent, so I'd stick with fader automation myself.

If you're wanting a little more detail and ambience to the sound, by
all means try the low-ratio, low-threshold compression you mentioned, as
this will usually work fine on most types of music. Don't stray over a
ratio of around 1.1:1 for classical recordings, though, if you want to
play things safe, and if you're getting gain-reduction of more than
about 4-5dB, you've probably got the threshold set too low. I'd
personally set the attack time fairly fast to track the signal levels
pretty closely, and then go for faster release times for more
detail/ambience and longer release times for less detail/ambience, but
this will inevitably be a matter of taste. Any isolated accented chords
will be particularly revealing of potentially unpleasant compression
artefacts, so listen out for how those sound.

You might be tempted to use multi-band compression with similar
settings, as many people do when working with more modern music styles,
but I'd steer clear of this, to be honest. The fluctuating tonal changes
that arise from this kind of processing are likely to upset the
delicate balance of the performance.

A more transparent approach to compression is to use a compressor as a
send effect, mixing the compressed signal in with the unprocessed one —
this is often referred to as parallel compression. For this to work,
you need to make sure that the compression processing doesn't also
introduce any delay, otherwise you'll get a nasty kind of static phasing
sound. That said, most software DAWs now have comprehensive plug-in
delay compensation, so this is becoming less of a problem for people
these days.

When working like this, you can usually get away with slightly
heavier compression, but I'd stay below a ratio of 1.3:1 to be on the
safe side. What some engineers do is automate the compressed channel's
fader, rather than the main channel's, adding in more of the compressed
signal during quieter sections. This can work really well, as it's often
when the music is quietest that it benefits most from added detail.

Monday, April 17, 2017

There are so many ways of connecting equipment
these days, such as S/PDIF, ADAT, AES-EBU and MADI, not forgetting good
old analogue. What are all the digital connections for?

SOS Forum post

Technical Editor Hugh Robjohns replies:
Back in the '80s and '90s, there were dozens of manufacturer-specific
digital interfaces, such as Yamaha Y1 and Y2, Melco, TDIF, ADAT, SDIF2,
R-Bus and many more, and none of them could be connected together. It
was a complete nightmare!

In order to make 'going digital' a
practical option, the Audio Engineering Society (AES) and the European
Broadcast Union (EBU) put their collective heads together and came up
with two generic, open-source digital interfaces: one for stereo and one
for multi-channel audio, the latter of which was originally intended to
link multitrack recorders to large consoles. These were called AES-EBU
(now more commonly referred to by the AES standards document number,
AES3), and MADI (Multi-channel Audio Digital Interface). A comparison chart showing different types of digital audio protocols.

AES3
was a bodge in the engineering sense, but the use of apparently
standard mic cables and connectors made it a familiar-looking interface
that reduced the fear and cost of 'going digital'.
The original
MADI specification essentially carried 56 channels, made up of 28 AES3
stereo pairs transmitted serially. A later revision called MADI-X
catered for 64 channels and is in widespread use today in applications
such as connecting stage boxes to digital desks, linking Outside
Broadcast trucks, and connecting the infrastructure in digital studio
complexes.

Today, AES3 is the preferred interface format for
professional stereo applications, although there is a noticeable trend
towards the AES3-id format which uses unbalanced BNC connectors and 75Ω
video cables rather than balanced XLR connectors and 110Ω cables.
AES3-id is a much better-engineered interface, and is far more
space-efficient. AES3 digits run with a fundamental frequency of 1.5MHz,
with strong harmonics all the way up to 10MHz and more. Video cable and
connectors are far better suited to handling those kinds of frequencies
than manky old mic cables, and AES3-id works more reliably over greater
distances, with less jitter as a result.

Having designed a very
versatile and effective digital interface, and all the hardware chips
to drive and receive it, Sony and Philips took the opportunity to use
the same thing for domestic applications and called it S/PDIF, with
coaxial (phono) and Toslink (optical) interfaces. The nitty gritty of
the auxiliary information and metadata carried by AES3 and S/PDIF are
slightly different, but the basic structure and audio formatting are
identical, and you can normally interconnect AES3 and S/PDIF with little
problem. S/PDIF is electrically almost identical to AES3-id.

In
terms of the actual interface properties, AES3 runs balanced signals
with a nominal 7V peak-to-peak swing, feeding a receiver with a minimum
sensitivity of 200mV. Because the signal starts so big, it tends to go a
long way (more than 100 metres) even on nasty mic cables. Put it into
decent low-capacitance 110Ω cable and it will easily travel 300 metres.
AES3-id is unbalanced and starts at 1V. The receiver sensitivity is the
same 200mV, while S/PDIF is also unbalanced and starts at about 0.5V.
The receiver sensitivity is also 200mV. The lower starting voltage is
why S/PDIF doesn't travel very far.

There are several
eight-channel AES3 interfaces, most using 25-pin D-Sub connectors.
Sadly, there are lots of different incompatible pin-outs: Yamaha,
Tascam, Genex and Euphonix, to name a few. But the Yamaha and Tascam
formats are the most prevalent.

Yet another variant of AES3 is
called AES42. This still uses XLRs and balanced cable, and the data is
encoded in exactly the same way as AES3, but it is intended for carrying
the output of digital microphones. The critical difference is that an
AES42 input socket provides 10 Volts of phantom power, and that power is
modulated in a specific way to allow remote control and digital
clocking of the microphone. It is an agreed format that has been adopted
by Neumann, Sennheiser, Schoeps and others, and will start becoming a
common feature on digital consoles and professional recording
interfaces.

In terms of other digital interfaces, Tascam's TDIF
is virtually dead, but ADAT is alive and well and in widespread use.
ADAT uses the same Toslink fibres and connectors as S/PDIF, but with a
different data-stream structure to carry eight channels.

In
addition to MADI for high numbers of channels, we also now have the new
AES50 SuperMAC and HyperMAC audio networking interfaces (originally
developed by Sony Oxford in the UK and now owned by Telex Communications
under the Klark Teknik brand).

SuperMAC provides 48 channels
bi-directionally over Cat 5 cable, while HyperMAC provides up to 384
channels bi-directionally over Cat 5 or Cat 6 or fibre. The signal
format includes embedded clocks in the same way that AES3 does.

Friday, April 14, 2017

The crack team of Paul White and Hugh Robjohns have
travelled the world solving readers' problems. Here, they down the Hob
Nobs and answer some of your recording queries in our Q&A
mini-series, Sound Advice.

Hugh: What we're
talking about here is sound arriving at different mics at different
times due to the different physical distances between the sound source
and the mics. Sound travels relatively slowly at around 340m/s (roughly
one foot per millisecond or one metre in three milliseconds). So if you
place one microphone a foot behind the other, the more distant mic will
capture that sound roughly 1ms after the closer one.

Paul:
In these situations, complex filtering occurs whereby some frequencies
are enhanced and others attenuated, depending on the exact time
difference between the two signals. A graph of such an affected waveform
shows lots of sharp peaks and troughs that look not unlike the teeth of
a comb (as shown below). Hence the term comb filtering is commonly
applied to the phenomenon. The filter notches and peaks are strongest
when the two signals are exactly the same level, and once you're
familiar with the rather coloured sound that comb filtering creates,
you'll always recognise it when you hear it again. Hugh:
So how can you prevent this from happening? Well, the simple answer is
not to use multiple mics on the same source in the first place, and to
minimise any spill from a loud sources that could reach several mics in
the room. If that is not physically possible, or if you deliberately
want to combine the outputs of two or more mics, then you need to be
prepared to spend some time optimising the combined sound. As Paul
mentioned, the comb filter effect is at its worst when the signal level
from the two (or more) mics is the same. So it helps if you make one
mic's contribution much less than the other's. That way you'll get some
of the tonal flavour, without nasty 'phasiness'. Something more than
10dB of level difference is typically needed, although you may get away
with less in some situations.

Paul:
Minimising spill between microphones is also key to keeping unwanted
phase effects at bay, so cardioid mics can sometimes be used in place of
omnis where their directional characteristics may be exploited to
reduce the amount of unwanted sound getting into the mic. Acoustic
screening between microphones, where practical, will also help and, in
some situations, such as with tom mics on drums, you can use gating to
mute the mics' signals, so they don't contribute the mix when not being
hit.

Hugh: You can also often improve the
situation by time-slipping one of the microphone signals inside your DAW
to re-establish time alignment. If you anticipate having to do this, it
is helpful if you have a timing reference to work with on the
recording. The simplest way is to record a sharp click at the start of
the take — a bit like the classic sync clapper board used in old film
shoots. Simply tap the instrument once with something to generate a
simple, clear click, wait a few seconds to make sure the reflections
from that sound have died down, and start the performance. When
everything is finished, it will then be easy to find that click at the
start of the appropriate tracks and use it as a reference marker when
time aligning the relevant tracks. Whether you slide the more distant
mic track forward, or the closer one back depends on how the overall
source timing works with the other tracks.

Paul:
Sometimes, the precision of time slipping tracks isn't necessary or
required, and a simpler approach will suffice. If you fade up two or
more channels, flipping the signal polarity of one with the preamps's
'phase' button will be sufficient to confirm whether that mic's
contribution is constructive to the mix or not. You will usually find
that there is obviously more bass in one signal polarity than the other ­
you would normally select the position that provides the most bass.

Hugh:
Most larger consoles incorporate a 'Phase Meter' to try to provide
warning of this situation, and most DAWs will have a similar facility
somewhere. Phase meters are scaled from +1, through zero and on to -1.
If the two channels are perfectly time-aligned and carrying identical
levels, the meter will show +1. Perfect mono! In general terms, when
listening to a normal stereo mix, anything on the positive side of zero
will produce acceptable mono without audible phasing problems. If the
needle (or LED marker) dips below zero and stays there, you have some
kind of timing/phasing problem, and listening in mono will reveal that.
It's not uncommon to find the phase meter dip briefly under the zero
mark — especially when there are complex reverbs involved — and as long
as it is only a brief dip towards -1, there is probably nothing to worry
about.

Tuesday, April 11, 2017

Is there any reason not to combine, say, an M/S
stereo mic pair with an additional spaced mic pair when recording things
like acoustic guitar? I've tried it and thought it sounded pretty
sweet. I'm pretty sure other people use this method on acoustic guitar,
even if it's not strictly by the rule book, but am I at risk of
confusing the stereo image?

SOS Forum postThere's
no reason why you can't use a spaced pair of mics alongside a
coincident pair, such as the Mid/Side arrangement shown here. But when
recording a single instrument it can have little effect, unless you are
specifically trying to capture a lot of ambience from a reverberant
room. Using the Mid/Side approach has advantages over a crossed pair, as
you can get a full, on-axis sound from the centre mic, with the sides
of the figure-of-eight adding some 'space' to the sound.

SOS contributor Mike Senior replies:
The bottom line with this is that your ears should be the guide. As far
as stereo imaging is concerned, I'm not sure there's enough width in an
acoustic guitar to give away what you're doing unless you get really
close with the microphones, and I find that extreme close-mic positions
often don't work very well with acoustic guitar.

I certainly
don't think you're transgressing against any real rules. For example,
the well-known Decca Tree method is quite similar to what you describe,
with the exception that you're using an M/S array rather than a single
omni mic in the centre. In fact, I understand that the original Decca
engineers also experimented with replacing their central mic with a
coincident stereo pair, and I can think of a good reason for doing so:
although the straight Decca Tree makes a lovely spacious sound with good
mono compatibility, its stereo imaging isn't as sharp as you'll get
from a coincident pair. Putting a coincident pair at the centre of a
Decca Tree adds sharper stereo imaging without impacting mono
compatibility.

Your choice of an M&S rig for this purpose has
a number of potential advantages over a crossed pair. With a single
instrument, a crossed-pair setup is likely to have the instrument
off-axis to both mics, and this will compromise the frequency response
at the centre of your stereo image, especially if you're using budget
mics. With an M&S setup, you can take advantage of the clearer
on-axis response of your Middle mic. Another potential advantage of
using M&S is that you can experiment with using an omni polar
pattern for the middle mic, which will give you a better low-end
response and more resistance to proximity effect if you're close-miking.
Summing to mono in this case would leave you with the same situation as
with a traditional three-mic Decca Tree, because the Sides signal from
the figure-of-eight mic completely cancels itself out.

Saturday, April 8, 2017

I recently purchased some Sennheiser HD600
headphones and have been told that I'll need a good headphone amplifier
to make the most of them. Currently, I use the headphones output of my
audio interface (a Digidesign M Box 2) to power my cans, but it seems to
distort at high levels. Do you think it would be advisable to get
myself a headphone amp? If so, how would I get the signal to the
headphone amp from my audio interface? I'm thinking of getting either
the Presonus HP4, Samson's C Que 8, the CME Matrix Y, or the Behringer
AMP800, mainly because the configuration of the inputs and outputs are
suitable for my system. It's probably worth mentioning that I use a
small Yamaha desktop mixer as a front-end to my DAW.

Michael Fearn

PC music specialist Martin Walker replies: Most
audio interfaces provide fairly clean-sounding headphone outputs,
although it can be tricky to predict how loud a particular set of phones
can go through a particular headphone amp without sounding strained.
According to the Digidesign web site, the M Box 2's headphone output can
provide six milliwatts into 50Ω, and since most such amps provide less
power into higher impedances such as the 300Ω of the Sennheiser HD600s, a
dedicated headphone amp might help you gain increased level while
helping your new headphones to sound as clean as possible.
Sennheiser's HD600 (right: reviewed in SOS June 2002) are
open-backed, reference-quality headphones. They are renowned for their
wide and spacious sound, but are at their best when used with a good
headphone amplifier, such as the Grace M902 pictured below.

However, the models you propose all have four headphone outputs,
which is fine if you need to plug in four pairs of headphones (so that
an entire band can monitor simultaneously, for example), but is not so
suitable if you simply want to get higher audio quality for your single
pair of Sennheisers. It sounds as though the latter is the case, so you
simply need to get hold of a single stereo headphone amp and connect it
between the stereo outputs of your audio interface and the inputs of
your Yamaha mixing desk. By spending your money on this one headphone
output you should be able to get better audio quality than spreading it
across four.

I say 'should', but, unfortunately, while there are lots of handy
multiple-output headphone amps at budget prices, once you start looking
for high-quality, single-output headphone amps the prices tend to shoot
up alarmingly. The Sennheiser HD600s are superb headphones, and some
people are prepared to spend over £1000 on a headphone amp to get the
very best out of them. Take the Grace M902 (www.gracedesign.com), which I mentioned in my January 2007 feature on headphone mixing (www.soundonsound.com/sos/jan07/articles/mixingheadphones.htm).
It costs £1400, but is far more than a rudimentary volume control.
Another widely recommended amp is Graham Slee's Monitor Class model,
which costs £475 (www.gspaudio.co.uk). Of course, you may still consider this 'silly money'.

I've scoured the Internet trying to find a reasonably priced
headphone amp to use with my Sennheiser HD650's, since it seems bizarre
to end up paying several times more than the cost of your audio
interface just to improve slightly on its integral headphone output. But
while there are plenty of audiophile products with prices to match,
there currently seem to be few products available to suit more modest
budgets.

One possibility is Creek's OBH21 (www.creekaudio.co.uk),
which retails at £190. I haven't auditioned it myself, but I know of
happy HD600/650 owners using this model. Another is the Rega Ear at
around £150 (www.rega.co.uk/html/ear_2001.htm), which, again, is used by many HD600/650 owners, while the Pro-Ject Head Box Mk 2 (www.sumikoaudio.net/project/products/headbox2.htm)
seems a bargain at £75 (yet still wins awards for its audio quality
compared to the average headphone socket found on a hi-fi amp), and will
deliver 60 milliwatts into 300Ω phones, or 330 milliwatts into 30Ω
models, which is a lot more than the M Box 2! What's more, it has
additional line-level outputs that you could connect to your mixer.

The Pro-Ject Head Box Mk 2 is available from local hi-fi shops or can
be bought on-line (in the UK) from various retailers, including
Noteworthy Audio (www.noteworthyaudio.co.uk), Stone Audio (www.stoneaudio.co.uk), or Superfi (www.superfi.co.uk). Readers in the USA seem to benefit from a much wider selection of indigenous products. Visit Head Room (www.headphone.com), and you'll find a huge range of headphone amps.

Thursday, April 6, 2017

The crack team of Paul White and Hugh Robjohns have traveled the world solving readers' problems. Here, they down the Hob
Nobs and answer some of your recording queries in our Q and A
mini-series, Sound Advice.

Hugh:
The djembe is a type of African drum, shaped like a goblet, and made
from a single piece of wood. Its head, traditionally goatskin, is
stretched over the top and tensioned with string. Like the conga, the
djembe is played by hand, and has an opening at the bottom of its
hollow, tubular shell which acts as a Helmholtz Resonator. Most of the
tone and volume of the bass resonance emerges from the bottom of the
tube when the drum is struck in the centre of the skin, while the
higher-pitched percussive sounds come from the drum skin when played
closer to the edge.

Paul: Because of that
large bass tube, the djembe needs to be lifted off the floor to to allow
sound to escape from the bottom of the drum. For recording, many
players use a metal floor stand or cradle, rather than holding it
between their knees, as it keeps the drum still. The pictures, right,
show a traditionally made djembe in such a stand.

Hugh:
When it comes to miking the drum, I'll always listen carefully to the
sound that the drum is making in the room. Only then will I make an
informed decision on where to place the mics.
If you have a nice-sounding room, and can afford to mic the drum from a distance, then a single (mono or
stereo) mic, positioned in a place where there's optimum balance
between bass and percussive sounds, should give good results. However,
in any kind of performance situation where there are other instruments
playing in the vicinity, or where you specifically want a closer, more
intimate sound, or where the room has no worthwhile sound character, you
are almost certainly going to have to use two close mics. This is
because the traditional approach of a single mic above the drum head
simply won't capture the bass energy in the right proportion.

Paul:
When tackling the top of the drum, I'd usually start around 12 inches
from the drum skin, using a small-diaphragm, cardioid capacitor
microphone. If you don't have one of these, a good dynamic will yield
good results, but you'll have to get closer to the drum head, maybe
within four to six inches.

Hugh: Whatever you
use and wherever you place it, try to aim it roughly halfway between the
edge and the centre to capture the percussive sound of the hand hitting
the head. You can experiment with the exact angle and aiming point of
the mic to change and control the tone slightly. The second mic is used
to capture the weight and tone of the bass resonance from below, and
would usually be placed somewhere close to (or maybe even an inch or two
inside) the mouth of the base, as shown in the pictures. Alternatively,
you could place a pressure zone (boundary layer) mic on the floor below
the mouth of the djembe, or use an omnidirectional mic on a thin sheet
of foam directly on the floor. This approach gives a more natural bass
sound, to my ears, and is well worth a try if you have suitable mics.
Just beware of foot-tapping or other floor vibrations. Of course,
experimentation is the key, because the best djembes are hand-made, and
every one will be different.

Paul: As is regularly mentioned in the hallowed pages of SOS,
if the room is adding unpleasant tonality to the sound of the drum,
hanging the ubiquitous duvet around and to the sides of the drum will
help damp down the sound. It is possible to over-dampen, so try sticking a tea tray under the drum if you think it's a little dull.

Hugh:
Mic selection is much the same as for any percussion or drum kit, hence
the use of the AKG D112 inside the base. (Alternatives include Shure's
Beta 52, the EV RE20, or Sennheiser's E902 — basically anything suitable
for miking bass drums). When miking the skin, good-quality dynamic mics
such as the venerable Shure SM57 or Sennheiser MD421 will tend to give a
slightly fatter, more flattering sound. A condenser mic such as a Rode
NT5 or AKG C451, by contrast, will tend to give more detail and
percussive edge, but watch out for overloads if you're close-miking. A
mic with a 10dB pad is often a good idea.

Paul:
Once you've got your djembe recorded, you may find that the levels vary
too much on playback, as the instrument is very dynamic. Compression
will help, and using an attack time of around 10ms will allow the drum
hits to come through cleanly, while taking around four to five decibels
off the very loudest peaks should help to even them out. You may have to
be more severe on recordings with excessive dynamic range.

Wednesday, April 5, 2017

I have been recording music for quite a few years
now, but I always get the same problem when I listen back to my mixes:
mine are far quieter than any commercial releases. Can you give me any
tips on how to get the sound louder without the track distorting? Or are
there any effects I can put on a track to boost the volume?

James Atkins

SOS contributor Mike Senior replies:
Maximising the loudness of final mixes is something of a contentious
issue, with different professional engineers holding widely divergent
views on the matter. Assuming that your mix is already peaking at 0dBFS,
increasing loudness beyond this is always a compromise, because you
inevitably have to change the recorded waveforms in some way. Whether
the compromises of any particular audio process are sufficiently offset
by the increase in loudness is a decision you have to make for yourself.The multi-band tube-compression processor in Drawmer's DC2476 Masterflow unit is simple but very effective.

The best tactic is simply to line up your mix in your DAW alongside a
couple of commercial mixes you rate, and process your mix to try to
bring it into line. The higher the quality of your monitoring
facilities, the better you'll be able to judge what you're doing. (With
mediocre monitoring, you'll tend to overdo the loudness because you
won't be sufficiently aware of the associated decreases in signal
quality.)

So what processing might you try? Well, the least deleterious change I
can think of making to increase loudness is to high-pass filter the mix
at a very low cut-off frequency (for an example, see the screenshot on
page 20). If you have any DC (0Hz) signal on your recorded tracks, this
can offset the whole audio waveform such that it clips earlier than it
should; high-pass filtering will remove this. You can also use the
high-pass filter to cut any low-frequency rumble you don't want, which
will give you more headroom to fade up the overall track level.

While we're on the subject of EQ, it's worth mentioning that the
human ear isn't as sensitive to high and low frequencies as it is to the
mid-range, and people tend to interpret brighter sounds in particular
as being louder. If you can detect a tonal difference between your track
and your references, I'd recommend trying gentle shelving EQ cuts to
match the sounds more closely in the first instance.

You might also want to have a look at plug-ins like Logic 's Match EQ and the TC Works Assimilator, or the stand-alone utility Harbal.
These all compare the frequency content of a reference track with that
of your own mix, and will suggest an EQ curve to match the two
automatically. Just be sure to take the suggested EQ curve with a pinch
of salt, as the automatic process is unlikely to be foolproof.

Subtle tape, valve, or transformer distortion processes can be a very
nifty way of increasing the subjective loudness with very little
increase in metered signal level. There are masses of software options
here, such as Silverspike's freeware Rubytube or the Magneto plug-in built into Cubase SX2.
Some nice hardware possibilities include the DRG process built into a
number of TC Electronic's rack processors, or the lovely multi-band tube
processing (shown on the left) in Drawmer's DC2476 Masterflow unit.

Compression can give comparatively transparent increases in loudness,
especially with low-ratio (below around 1.3:1) and low-threshold
(between maybe -30dBFS and -50dBFS) settings. Full-band compressors will
be more tonally transparent in this role, but multi-band models can
usually add more loudness without pumping artifacts, albeit with the
side-effect of dynamically tampering with the tonality of the mix.

For rock and dance styles, full-band compression can be used with
higher threshold and ratio settings to create an illusion of extra
loudness through pumping compression effects. If you want to experiment
with this, try starting with a 2:1 ratio, a 1ms attack time and about
100ms of release time. If you then set the threshold level so that the
compressor reduces the gain mainly on drum hits, you should hear a
pumping effect, and you can adjust the ratio and release time to
regulate the strength of it.The
zoomed-in waveform of a kick-drum hit from Dr Dre's 'The Watcher', from
the album 2001, showing that the track has been clipped to increase its
loudness. The highlighted region in this screenshot clips around 140
consecutive samples.

If you find that compression pumping
is knocking the bass frequencies out of your kick drums, you may want to
increase the compressor's attack time to let more of the sound through
before the compressor clamps down. Alternatively, you could high-pass
filter the compressor's side-chain to decrease the gain-reduction
element's sensitivity to low frequencies.

Limiters are often used to give a loudness increase, and can manage a
comparatively loss-free increase of several decibels in the right
circumstances. There are now lots of full-band and multi-band models
available, but one side-effect I'm not fond of with any of them is that
they tend to cause any heavy drum sounds in your track to lose punch and
to sink back into the mix. For less percussive material, the trade-offs
of processing depend on how you the limiter's release time is set, but
the main things to listen for are unwanted pumping effects and bass
distortion.

If you've tried all of the above and you're still short, then maybe
it's time to zoom in on the waveforms of your reference tracks to see
whether they've been clipped. Although clipping is frowned on by many
engineers, the practical reality is that clipping is all over commercial
releases in many styles, so you need to give the issue of clipping
serious consideration in order to decide where you stand. The main sonic
disadvantage of clipping is a type of distortion which in itself is
pretty unmusical. However, many engineers take the view that a degree of
clipping can be well enough disguised under certain circumstances that
it becomes a reasonable price to pay for a considerable loudness hike.

The first main situation where it's often used is when the track has a
lot of fuzzy sounds, such as distorted electric guitars, into which the
clipping artifacts can fairly easily blend. Some stark examples of this
are The Darkness' 'Growing On Me', Chemical Brothers 'Block Rockin
Beats', and Pink's 'Feel Good Time', to name a few.

The other common use of clipping is in music styles that are very
drum-heavy, such as rock and hip-hop, and here the technique is to clip
mainly just the drum beats. Unless you drive things too far, the burst
of distortion on each hit tends to be perceived mostly as an alteration
in the tonality of the drum hits, rather than as distortion per se, so many producers are happy to take advantage of that. To take one example, Dr Dre's 2001
album frequently clips in excess of 100 consecutive samples on its
kick-drum hits (see the screenshot on page 22), and that kind of
clipping is by no means out of the ordinary in a lot of modern
commercial music styles.

The only thing I would say is that no matter what you do, you should
always make a habit of retaining a completely unprocessed version of
your mix file for safety's sake. Not only will this mean that you don't
burn any bridges if you decide at a later date that you mucked up your
settings, but it also means that you can always still take your material
to a professional mastering house if you want. Their specialist
equipment and engineers will be able to get much better (and louder)
results from an unprocessed file than from a processed one.